Web Management Interface for Kamailio (OpenSER) SIP Server

Posts Tagged ‘kamailio’

A new E-Learning class about SIP Router Configuration File is due to start on Oct 4, 2010. Registrations are accepted up to Oct 2, 2010.

The class duration is three months and gives the opportunity to learn the structure of configuration file and how to write it properly. Lessons are applied to Kamailio (OpenSER) and SIP Router SIP servers, touching VoIP security and scalability, at a fee of just several consultancy hours.

Next class Kamailio Advanced for Carriers, organized by Asipto in collaboration with IPCom Network and NAP of the Americas, focuses on the features that allow you to build a carrier SIP infrastructure:

Next SIP Router Masterclass will be held March 22-26, 2010 in Berlin, Germany.

Teachers:

Daniel-Constantin Mierla – co-founder of Kamailio (former OpenSER) project in 2005, currently core-developer and member of project’s management board

Olle Johansson – Asterisk developer and member of the Digium Asterisk Advisory Board.

End of 2008, Kamailio (OpenSER) and SIP Express Router (SER) started a joint collaboration under http://sip-router.org project, bringing together valuable developers and architects of SIP servers. Kamailio (OpenSER) is now at version 3.0.0 (released on January 11, 2010), being based on SIP-Router.org project.

Learning to configure the SIP server is not easy, but is the key for a successful and secure VoIP business. The flexibility of SIP routing engine allows you to implement in no time innovative services, IP telephony, Instant Messaging, Presence and beyond. Asterisk comes to complete with rich media services and applications. Doing everything designed right and scalable saves time and money.

We create the opportunity for you, guided by experienced instructors, to learn how to build an Unified Communication platform from scratch using the SIP server engine and Asterisk.

Kamailio (OpenSER) and SER are the leading open source SIP servers, routing billions of minutes and handling millions of active VoIP users each month. Well known for its stability and flexibility the SIP Express Router (SER) family of SIP servers is continuously increasing the adoption on the market. With the launch of SIP-Router.org project, the SIP server secured a reliable development team, backed up by strong business community.

A new E-Learning class about SIP Router Configuration File is due to start on February 8, 2010. Registrations are accepted up to February 5, 2010.

The class duration is six months and gives the opportunity to learn how to write proper configuration files for SIP Router and Kamailio (OpenSER) SIP servers, how to take care of VoIP security and scale the service.

Target attendees:

VoIP administrators willing to learn and have a support channel for SIP Router configuration file

System and network administrators willing to enhance their portfolio with VoIP knowledge

It is a major release of Kamailio (OpenSER), following ten months of development and heavy testing. Represents a special release, being based on SIP Router project, therefore enables admins to blend Kamailio (OpenSER) and SIP Express Router (SER) core features and modules in same configuration file.

Given the above, this version brings out an impressive number of new features and improvements, read the release notes at:

Next SIP Router Masterclass will be held November 9-13, 2009 in Berlin, Germany.

Teachers:

Daniel-Constantin Mierla – co-founder of Kamailio (former OpenSER) project in 2005, currently core-developer and member of project’s management board

Olle Johansson – Asterisk developer and member of the Digium Asterisk Advisory Board.

By end of 2008, Kamailio (OpenSER) and SIP Express Router (SER) started a joint collaboration under http://sip-router.org project, bringing together valuable developers and architects of SIP servers. Kamailio 3.0 and SER 3.0 (to be released soon) become compatible in terms of configuration file and extensions.

Learning to configure the SIP server is not easy, but is the key for a successful and secure VoIP business. The flexibility of SIP routing engine allows you to implement in no time innovative services, IP telephony, Instant Messaging, Presence and beyond. Asterisk comes to complete with rich media services and applications. Doing everything designed right and scalable saves time and money.

We create the opportunity for you, guided by experienced instructors, to learn how to build an Unified Communication platform from scratch using the SIP server engine and Asterisk.